Fixed and mobile communication terminals have so far been used mainly for making voice calls. Standardised and well-working communication technologies and protocols are then utilised to communicate voice between fixed and/or mobile terminals using circuit-switched communication channels. In particular, radio based circuit-switched channels for mobile terminals have been designed and optimised to provide acceptable quality and reliability for voice calls, at the same time requiring a minimum of bandwidth in order to increase network capacity.
A multitude of new telephony services are now rapidly being developed which can be employed in particular by the introduction of new technologies allowing notably higher transmission rates and increased network capacity. For example, GPRS (General Packet Radio Service) and WCDMA (Wideband Code Division Multiple Access) technologies are currently emerging for enabling wireless telephony services requiring a wide range of transmission rates and different protocols. The trend today is also a move towards packet-switched networks and technologies providing more capacity and flexibility as compared to the traditional circuit-switched networks. Further, new sophisticated mobile terminals are also emerging on the market, equipped with functionality to handle the new services.
Many of these new services involve real-time transmission of video information as well as audio information, and may further involve the transmission of added data representing text, documents, images, audio files and video files in a multitude of different formats and combinations. Such services are generally referred to as “multimedia” services, which term will be used in this description to represent any telephony services that involve the transfer of any data in addition to ordinary voice.
A prevailing goal or ambition is to converge all services on to a single transport mechanism—the packet based Internet Protocol (IP), regardless of the type of access networks and technologies. Recently, a network architecture called “IP Multimedia Subsystem” (IMS) has been developed by the 3rd Generation Partnership Project (3GPP) as an open standard, to give operators of access networks the ability to offer multimedia services in the packet domain. An IMS network, comprising various different network elements to handle the services, can be built above any type of access network and is more or less independent of the access technology used, provided that the access network is able to support the service requirements of IMS in terms of bandwidth, QoS (Quality of Service), etc. Hence, IMS is a platform for enabling services based on IP transport, and is basically not restricted to any limited set of specific services.
However, the packet based IP transport technology is currently not quite suitable for voice communication mainly due to shortcomings in quality and reliability. In short, the difference is that a circuit switched channel is a permanent connection for the duration of a call with a fixed and guaranteed bandwidth, resulting in fairly consistent quality and reliability, whereas in packet switching a connection of variable bandwidth is temporarily established whenever there are any packets to transmit. Packet switching is therefore inherently associated with various unpredictable transmission delays and packet losses that may potentially result in unacceptable variations of quality, bitrate and reliability. In particular, a radio link is typically the critical part of a transmission path due to its limited bandwidth. Transmission delays can therefore be a significant problem for packet-switched radio channels.
As a result, a traditional circuit switched (duplex) voice bearer is currently considered to be better than a packet switched (non-duplex) voice bearer in this respect. Hence, network operators are not yet able to launch all-IP multimedia services involving voice transport with full duplex, in 3G mobile networks.
To overcome these problems, it has been proposed that a multimedia communication session should be divided into a circuit-switched part for the voice transport using a circuit-switched telephony system, and a packet-switched part for the transport of other data using a packet-switched telephony system based on IP technology. According to this proposal, circuit-switched bearers are used for voice, particularly in radio links, whereas completely separate packet-switched bearers are used for other medias. In this way, the high performance associated with the traditional full duplex voice channels is obtained, whereas any other data involved in multimedia services can be adequately supported by packet-switched transport, since it is normally not equally delay-sensitive. This arrangement can also reduce the costs for network operators by utilising existing resources for circuit-switched transmission, as e.g. in GPRS networks having both capabilities.
This solution is schematically illustrated in FIG. 1 where two mobile terminals A and B are engaged in a multimedia communication involving both voice and data. The terminals are connected to access networks, not shown, providing radio access, as schematically illustrated by blocks 100A and 100B, over respective radio channels. Here, it is assumed that each access network has separate architectures and logic systems for circuit-switched and packet-switched transport, respectively.
According to this solution, the communication flow through the various networks and nodes involved in the session between the terminals A and B is divided into a circuit-switched (CS) part and a packet-switched (PS) part. The CS part is transported over a separate circuit-switched logic system, as represented by the block “CS logic” 102. Any other data involved with the used multimedia service, such as video, images, text, etc., which will be called the “PS part” for short hereafter, is transported over a separate packet-switched logic system, as represented by the block “PS logic” 104. It should be noted that FIG. 1 does not show any specific networks or nodes, but simply illustrates how the communication flows are handled logically.
However, a considerable drawback with this solution is that each logic system has its own network service control function. This is schematically illustrated in FIG. 1 as a session control unit “SCU” 102a supporting the circuit-switched part of the session, and another session control unit “SCU” 104a supporting the packet-switched part of the session. For example, the session control unit 104a, may reside within an IMS network as described above and handles the data transport, whereas the session control unit 102a resides within a circuit-switched network and handles the voice transport separately. As a result, different session events occurring in the CS part and the PS part, respectively, will be noticed and handled in isolation from the other part.
For example, the PS part of the communication will be unaware if the CS part is terminated due to, e.g., intentional on non-intentional disconnection of a voice call. In response thereto, it may be desirable that the PS part is then also terminated, or that any other action is taken in the PS part, depending on the nature of the service used. Moreover, the billing function is handled separately by the two system parts, and therefore any common charging and discounting of such multimedia services requires that the two billing functions must somehow be coordinated, e.g. by means of a common billing center or the like.
The problems associated with the control functions having separate session control units are naturally avoided if both voice and data are transported over the same packet-switched logic, as illustrated in FIG. 2, where the PS logic 202 is controlled by means of a single session control unit SCU 202a, e.g. within the context of IMS. However, the above-described problems with voice communication over IP will then remain unresolved.